Issues 05000 - 05999

[..]
ASTERISK-05000: rtptimeout counts when using app_record.
ASTERISK-05001: [patch] Change debug messages to explain what's going on with RTP and NAT
ASTERISK-05002: [patch] Add new options to README.asterisk.conf
ASTERISK-05003: [patch] Janitor task: Move from ast_exists_extension to ast_goto_if_exists
ASTERISK-05004: [patch] [post 1.2] safe asterisk restart script to be used by unix illiterates
ASTERISK-05005: Does not conform to 1.2 style behaviour with regards to 'j' jump flag.
ASTERISK-05006: overlapping ringing and voice
ASTERISK-05007: [patch] Fix silly ERROR messages in chan_sip
ASTERISK-05008: [patch][post 1.4] agents in realtime
ASTERISK-05009: [patch] [post 1.2] Switch indentation
ASTERISK-05010: [patch] app_while first loop executes even if evaluates to "0"
ASTERISK-05011: [PATCH] Transfer app never reports TRANSFERSTATUS = FAILURE
ASTERISK-05012: [patch] Small speed improvement to ast_rtp_raw_write
ASTERISK-05013: After 2 Sep 2005 update of iax2-parser cannot receive incoming iax2 connections with rsa authentication
ASTERISK-05014: [request] [post 1.2] - Implement "hintexten" sip peer setting
ASTERISK-05015: Crashes with nocdr after resetcdr in dial plan
ASTERISK-05016: [patch] acl.c : IPTOS_{LOW,MIN}COST not defined in NetBSD
ASTERISK-05017: IAX2 REGREL causes INVAL response
ASTERISK-05018: [patch] Allow res_features application mapping to work
ASTERISK-05019: Fix ringing after 100 Trying introduced in #5127
ASTERISK-05020: [patch] General fixes to make CVS-HEAD compile on NetBSD and FreeBSD, possibly others.
ASTERISK-05021: Cisco SIP (g729a) to Cisco SIP (g711ulaw) bridge results in Asterisk Crash
ASTERISK-05022: man pages should not be compressed
ASTERISK-05023: [patch] OSP authentication issue
ASTERISK-05024: Asterisk does not recognizes DTMF when is dialed TOO FAST and numbers are repeated. Using Cisco DTMF (type 121)
ASTERISK-05025: [patch][post 1.4] Manager Event Hook
ASTERISK-05026: [branch] RTP Packetization
ASTERISK-05027: app_queue with realtime don't send calls to an recently added member to the queue_member_table
ASTERISK-05028: "Presence" subscription causes Internal Server Error on Polycom 600
ASTERISK-05029: app_queue with realtime sent two simultaneous calls to an already busy agent when 2 boxes are working together
ASTERISK-05030: Don't send BYE out of dialog
ASTERISK-05031: after the second register happens from asterisk 2 to asterisk 1 asterisk 1 cannot call asterisk 2, authentication error
ASTERISK-05032: [patch] Support OSP Toolkit 3.3.3
ASTERISK-05033: make valgrind fails on os x
ASTERISK-05034: Build failure due to missing ztcfg.sgml
ASTERISK-05035: Add ability for call files to run an application on success, failure and retry
ASTERISK-05036: [patch] Make rtp handling a bit more stable
ASTERISK-05037: [patch] Dial parsing error?
ASTERISK-05038: More documentation (doxygen)
ASTERISK-05039: Voicemail hangs up on you if you select an empty folder
ASTERISK-05040: IAX Hostname Resolving
ASTERISK-05041: [request] [post 1.2] Mismatch peers with same IP
ASTERISK-05042: inserting of install commands in modprobe.conf to automatically run ztcfg considered harmful
ASTERISK-05043: [patch] Alarm on on idle PRI span that recovers quickly will likely leave the PRI D-Channel stuck in "Down" status
ASTERISK-05044: Random Crashs on CVS HEAD
ASTERISK-05045: Asterisk does not send "pong"
ASTERISK-05046: [patch] [post 1.2] Inline backtraces
ASTERISK-05047: [patch] Add OSP token format option
ASTERISK-05048: [request] RFC3389 Support requested. (workaround)
ASTERISK-05049: 4-port ATA fails to register because * doesn't track nonce-value per line
ASTERISK-05050: [patch] Zaptel idles channels with a value that isn't appropriate for alaw
ASTERISK-05051: [patch] [post 1.2] Add HANGUPCAUSE to cdr engine
ASTERISK-05052: Bad file descriptor
ASTERISK-05053: [request] [post 1.2] ASTCC make another call without hangup
ASTERISK-05054: [patch] chan_zap.c not correctly handling polarity reversals caused by calling party hangups
ASTERISK-05055: man asterisk makefile failes to install 16 sounds.
ASTERISK-05056: safe_asterisk is installed w/ no execute permissions
ASTERISK-05057: asterisk-sounds package sounds-extras.txt has definitions with no .gsm extension
ASTERISK-05058: Asterisk does not obey rport parameter
ASTERISK-05059: coredump with libpri/asterisk
ASTERISK-05060: asterisk silently stops accepting IAX calls
ASTERISK-05061: [patch] Do not send 200 before 404 on subscriptions
ASTERISK-05062: user events are sent with EVENT_FLAG_CALL instead of EVENT_FLAG_USER
ASTERISK-05063: [patch] ENUMLOOKUP function for enhanced ENUM support
ASTERISK-05064: [patch] [post 1.2] Hangupcause to cdr_odbc + minor changes
ASTERISK-05065: PRI zap channels not cleared when no match in context for dialed number on inbound call
ASTERISK-05066: Build fails
ASTERISK-05067: zap show channels does not show phone number on incoming calls
ASTERISK-05068: [PATCH] [post 1.2] allow pri show span to show more spans
ASTERISK-05069: [patch] [post 1.2] allow safe_asterisk to set asterisk priority and adjust max open files with ulimit -n
ASTERISK-05070: [patch] SayAlpha does not read special characteres such as "," and "/"
ASTERISK-05071: progressinband fix
ASTERISK-05072: Asterisk crashes with any Sipura ATA connected to AMD64 system
ASTERISK-05073: [request] [post 1.2] DisplayName application
ASTERISK-05074: [patch] [post 1.2] System Load Average Monitor
ASTERISK-05075: ${AGENTBYCALLERID_${CALLERID} isn't updated when a logged in agent logs in again
ASTERISK-05076: Unknown SDP media type in offer: video 17374 RTP/AVP 31 34
ASTERISK-05077: [patch] Provisional responses to INVITE are ignored if a request has been sent in an early dialog
ASTERISK-05078: Message-Account needs to be after Messages-Waiting per RFC3842
ASTERISK-05079: The sdp-anat option tag is defined in RFC 4092
ASTERISK-05080: Calls to Voicemail via IAX drop (ast_play_and_record: No audio available)
ASTERISK-05081: ISDN Early Media does not work with CVS HEAD
ASTERISK-05082: [patch] Fix for registration and subscriptions
ASTERISK-05083: [patch] [post 1.2] Modified gotoiftime to add a branch if not in time range
ASTERISK-05084: SDP Error Message which isn't really an error
ASTERISK-05085: [patch] Incorrect syntax for ENUMLOOKUP function in docs/README.enum.
ASTERISK-05086: reg->messages can equal -1 in function iax2_ack_registry
ASTERISK-05087: [patch] Use ast_channel_setwhentohangup instead of accessing data structure directly
ASTERISK-05088: [patch][post-1.2] add CallerID, CallerIDName to OriginateFailure/OriginateSuccess
ASTERISK-05089: [patch] add CallerID, CallerIDName to OriginateFailure/OriginateSuccess
ASTERISK-05090: [patch] README.enum documentation examples repair
ASTERISK-05091: sip.conf spelling error
ASTERISK-05092: app_directory reports "No compatible entries" when there are
ASTERISK-05093: on Big Endian platforms, expressions $[] don't work with the new expression parser
ASTERISK-05094: revision 1.82 of logger.c is broken
ASTERISK-05095: LoggedinChan is not displayed via Manager interface.
ASTERISK-05096: ztdummy throwing error on kernel 2.6.13.1
ASTERISK-05097: [patch] [post 1.2] Allow filename to be optionally passed
ASTERISK-05098: Asterisk leaves zombie agi processes when running under linux 2.6
ASTERISK-05099: call forwarding to 4 digit extensions does not work
ASTERISK-05100: [branch] show managers + show manager foobar
ASTERISK-05101: [patch] [post 1.2] GET DATA now supports escape digits
ASTERISK-05102: [patch] Missing newline in debugging code
ASTERISK-05103: [patch] Error in README.enum example
ASTERISK-05104: AEL parsing issues on app_cut
ASTERISK-05105: EINTR is not checked on get_input at manager.c after poll, cause manager to disconnect
ASTERISK-05106: Call to ChanIsAvail fails
ASTERISK-05107: [patch] Curl ought to be a function
ASTERISK-05108: [patch][post 1.2] 802.1p support via SO_Priority in RTP.c
ASTERISK-05109: asterisk stops when trying to restart it
ASTERISK-05110: L option in Dial application does not disconnect
ASTERISK-05111: asterisk expressions doesn't work on os x
ASTERISK-05112: agent wrapuptime not working across queues
ASTERISK-05113: no jump to n+101 when busy
ASTERISK-05114: [request] alsa.conf needs a option for allowing non-blocking
ASTERISK-05115: 486 is now congestion rather than busy in sip_chan.c version 1.841
ASTERISK-05116: app_voicemail.c:5606 vm_box_exists: VM box ...@default exists, but extension ..., priority ... doesn't exist
ASTERISK-05117: Forwarding voice message prompts saved message in addition to forwarding voice message
ASTERISK-05118: IAX sets CALLERPRES to NOT AVAILABLE if no number is in CALLERID
ASTERISK-05119: [patch] [post 1.2] PostgreSQL driver for Asterisk RealTime (res_config_pgsql)
ASTERISK-05120: [patch] Invalid ast_verbose logged messages longer than 4K
ASTERISK-05121: [patch] ast_strlen_zero on a null string causes segfault.
ASTERISK-05122: [patch] [post 1.2] More explicit debug info for PRIs
ASTERISK-05123: Is impossible send dtmf digits to some toll free remote ivr using early audio
ASTERISK-05124: [patch] Add "UNDEFINED" Call-ID support for OSP module
ASTERISK-05125: [patch] SAY PHONETIC doesnt work in AGI
ASTERISK-05126: IAX realtime fails if auth is required
ASTERISK-05127: app_pickup deadlocks asterisk
ASTERISK-05128: [request] zaptel should allow hardware to handle HDLC
ASTERISK-05129: Can't fetch sip registration binding
ASTERISK-05130: [patch] [post 1.2] Addition of multiple configurable periodic announcements
ASTERISK-05131: [patch] MeetMe doesn't recreate pseudo when Local channel masquerades back to non-Zap channel
ASTERISK-05132: [patch] GetVar breaks parameter schema
ASTERISK-05133: [patch] voice mail cuts off on freebsd in agi
ASTERISK-05134: [patch] Danish time is said wrong
ASTERISK-05135: Mantis configuration error
ASTERISK-05136: Queues with periodic announce does not work
ASTERISK-05137: Event: Hangup gives garbled Channel Name
ASTERISK-05138: hints just shows states 0 and 4 on SIP devices
ASTERISK-05139: Cut and paste error in chan_zap when ZT_EVENT_NOALARM received?
ASTERISK-05140: DTMF is not recongnized in AGI
ASTERISK-05141: [patch] Cannot do substrings with functions
ASTERISK-05142: [patch] Result of EVAL is unterminated
ASTERISK-05143: Meetme Admin * menu
ASTERISK-05144: [patch] [post 1.2] Log app
ASTERISK-05145: libpri clocking errors
ASTERISK-05146: [patch] SIP RFC minor issue
ASTERISK-05147: [patch] Fix build problem on FreeBSD for CVS HEAD
ASTERISK-05148: [patch] libpri Makefile tweaks
ASTERISK-05149: [patch] Asterisk Makefile tweaks
ASTERISK-05150: [patch] Asterisk -p command line option doesn't work on Linux
ASTERISK-05151: cdr_addon_mysql Errors on each CDR record
ASTERISK-05152: only one way text using agents/queues
ASTERISK-05153: [patch] [post 1.2] New application Vars2File
ASTERISK-05154: [patch] poll.o/dlfcn.o no longer needed for MacOSX Tiger
ASTERISK-05155: Crashes while reading config files
ASTERISK-05156: [patch] enumlookup gives a fatal error during build
ASTERISK-05157: [patch] Asterisk segfaults with RealTime queues
ASTERISK-05158: some remark for 0002471
ASTERISK-05159: [patch] PRIREDIRECTREASON typo in chan_zap.c
ASTERISK-05160: Incorrect rtp protocol map for telephone event for native bridge between 2 SIP devices.
ASTERISK-05161: zt_handle_event: Ring/Off-hook in strange state 6 on channel X
ASTERISK-05162: Deadlocks - manager?
ASTERISK-05163: Call jumps into Digital Receptionist and is then dropped
ASTERISK-05164: [patch] no update of use count and possible memory leak in app_chanisavail
ASTERISK-05165: [patch] Wrong auth on BYE and no update of noncecount with qop digest auth
ASTERISK-05166: [request] [post 1.2] asterisk does not process priorities in DNS SRV records
ASTERISK-05167: [patch] [post 1.2] Add support for queue like functionality to astobj.h
ASTERISK-05168: Changed voicemail password doesn't stick if using Realtime
ASTERISK-05169: [patch] Recent conversion to ast_goto_if_exists breaks queue exit
ASTERISK-05170: Hardware HDLC in Zaptel
ASTERISK-05171: [patch] [post 1.2] SipGetPeerData
ASTERISK-05172: Intermitant delays on call setup.
ASTERISK-05173: [patch] zaptel.c has unchecked calls to copy_from_user() and copy_to_user()
ASTERISK-05174: [patch] [post 1.2] customizable number in default message
ASTERISK-05175: [patch] Asterisk checks for remote connection as 'rasterisk' but does not install rasterisk binary
ASTERISK-05176: priority in zt_bridge main loop doesn't swap
ASTERISK-05177: joinempty=strict have no effect
ASTERISK-05178: No sound when using AGI script Meetme
ASTERISK-05179: Recent RPID patch wasn't commited with the build_rpid in the right location.
ASTERISK-05180: DTMF on EM_WINK circuits isn't processed correctly in chan_zap.
ASTERISK-05181: [patch] Manager SendEvent
ASTERISK-05182: [patch] The SIP channel prevents placing non-telco phone numbers in the caller-ID number information.
ASTERISK-05183: if...else executes both branches
ASTERISK-05184: [patch] [post 1.2] FILTER dialplan function
ASTERISK-05185: [patch] make -j doesn't work
ASTERISK-05186: [patch] Remove extra space from authentication
ASTERISK-05187: [patch] Fix SIP timer t1
ASTERISK-05188: [patch] pridialplan and prilocaldialplan do not provide adequate control for all telco connections.
ASTERISK-05189: ChanSpy crash
ASTERISK-05190: IAX deadlock
ASTERISK-05191: Static Members - Retry parameter is ignored
ASTERISK-05192: [patch] HasVoicemail and HasNewVoicemail applications should be functions
ASTERISK-05193: retrans_pkt: Maximum retries exceeded on transmission XXXX for seqno 102 (Critical Response)
ASTERISK-05194: Makefile detection of $MODCONF is broken, causes failure during make install
ASTERISK-05195: [patch] ChanIsAvail CVSHEAD not functining with option 's'
ASTERISK-05196: CDR using the CALLERIDNUM for Outbound SRC - Fraud Possiblity when setting CallerID to "Private <>"
ASTERISK-05197: Consultative transfers let * crash (malloc() problem)
ASTERISK-05198: When only static members exists app i exiting with error.
ASTERISK-05199: option r does not generate ringtones when multiple interfaces are called
ASTERISK-05200: [patch] [post 1.2] Russian language in app_voicemail
ASTERISK-05201: [patch] [post 1.2] Say formatted date in Russian
ASTERISK-05202: [patch] [post 1.2] Allow min expiry in sip.conf
ASTERISK-05203: [patch] Updated OSP support in chan_sip and res_osp
ASTERISK-05204: Avoid transcoding by smarter codec negotiation with peers
ASTERISK-05205: Increasing/decreasing conference volume kills asterisk.
ASTERISK-05206: [PATCH] Update logger to report thread id instead of process id
ASTERISK-05207: [patch] Manager connection remains open after Logoff
ASTERISK-05208: [patch] quick clarification for voicemail
ASTERISK-05209: Manager disconnects slow client too quickly
ASTERISK-05210: Asterisk crash on a high load of SIP connections ( ~ 240 )
ASTERISK-05211: [patch] Switch default case not working inside a Macro
ASTERISK-05212: Manager connections randomly disconnected
ASTERISK-05213: Getting the clocking right for E1 and T1 lines
ASTERISK-05214: [patch] Update sip.conf.sample
ASTERISK-05215: [patch] Remove peer mask
ASTERISK-05216: [patch] Choppy dial tone on Linksys routers / FPU usage in indications.c
ASTERISK-05217: Asterisk sending incorrect facilities to qsig switch in network mode
ASTERISK-05218: [patch] [post 1.2] Expand fullcontact length
ASTERISK-05219: Asterisk 1.2b1 crash on Local channel hangup
ASTERISK-05220: crash about every 2 hours
ASTERISK-05221: Wrapup time not working properly
ASTERISK-05222: [patch] Tone definitions in indications.conf not properly validated
ASTERISK-05223: Privacy (rfc3325) feature in sip
ASTERISK-05224: Action: Logoff does not close port
ASTERISK-05225: [patch] Fix subscriptions for eye-beam
ASTERISK-05226: [patch] Incorrect parsing of IF, SWITCH and WHILE operators argument.
ASTERISK-05227: [patch] Wrap sample configs at 80 characters
ASTERISK-05228: [patch] [post 1.2] Asterisk Messaging with Dialog Emulation Support
ASTERISK-05229: The API Manager continue connection after a "logoff" Action
ASTERISK-05230: [patch] Asynchronous generation of outgoing frames when timing device available
ASTERISK-05231: 1.2.0beta1 ISDN strong&weird noise
ASTERISK-05232: haveing native MoH run in the background
ASTERISK-05233: Non-working authentication with Audiocodes MP-124
ASTERISK-05234: H323 channel dial-in freezes Asterisk
ASTERISK-05235: [patch] Fix a rather large memory leak in Directory
ASTERISK-05236: [patch] remove the "stale nonce" message
ASTERISK-05237: [patch] [post 1.2] Allow ${VM_CATEGORY} to be used in emails
ASTERISK-05238: Record File with DTMF termination leaves a bit of DTMF in the recording
ASTERISK-05239: When T38 re-INVITE is rejected with 488, RTP stream does not resume previous codec.
ASTERISK-05240: [patch] brackets inside a comment in voicemail.conf can cause an uninteded context change when changing password
ASTERISK-05241: [patch] Sort alphabetiquely the dial options
ASTERISK-05242: CVS btp (chan_btp) update due to update in libbluetooth1-dev (debian/2.19-1)
ASTERISK-05243: [patch] TOUCH_MONITOR filename not working
ASTERISK-05244: [chan_sip/app_dial?] CANCEL sent after 30 seconds when timeout >30
ASTERISK-05245: [patch] LANGUAGE variable not used in PlayBack/BackGround?
ASTERISK-05246: [patch] [post 1-2] Add the video stream for AGI function STREAM FILE
ASTERISK-05247: Voicemail ODBC is not reading envelop information from database
ASTERISK-05248: Enhancement to Dial to return hangup cause when unable to complete call
ASTERISK-05249: [patch] [post-1.2] Enabling pause and resume while monitoring a channel
ASTERISK-05250: [patch] New application to define Short Numbers in a context
ASTERISK-05251: [patch] Update for AMI's originate command's documentation
ASTERISK-05252: If cdr_mysql activated and connection do ds is lost the channels never hang-up
ASTERISK-05253: [patch] Remove superfluous checks in chanvars.c
ASTERISK-05254: [patch] When used with CVS-HEAD, ASTCC might leave the "inuse" flag for the card
ASTERISK-05255: [patch] Reconfigure parking extension properly (remove old extension)
ASTERISK-05256: segfault during atxfer
ASTERISK-05257: [patch]Indentation fixes.
ASTERISK-05258: Forced Redirect via Manager API causes * to crash
ASTERISK-05259: chan_sip doesn't form the SIP URI correctly when placing outgoing SIP calls received from another * box with IAX
ASTERISK-05260: [patch] usage of call files for outbound calls causes memory leak
ASTERISK-05261: [PATCH] Semicolons in call spool files
ASTERISK-05262: realtime ex-girlfriend dialplan matching problem
ASTERISK-05263: [patch] [post 1.2] Prelude to Comfort Noise Generation on Asterisk
ASTERISK-05264: Compressed manpages in source tarball
ASTERISK-05265: It would be nice to have the wW options on Queue
ASTERISK-05266: SIP REGISTER fails with unknown domain when no domain have been specified
ASTERISK-05267: [patch] [branch] Secure RTP (SRTP)
ASTERISK-05268: When internet-connection is broken and asterisk tries to register the whole sip-system fails
ASTERISK-05269: Does open source PBX work with any phones or just computer?
ASTERISK-05270: [patch] ? or <tab> doesn't work for top level CLI commands
ASTERISK-05271: Monitor fails after transfering call
ASTERISK-05272: Changes between 1.2-beta1 and latest CVS HEAD make all SIP calls from PROXY fail..
ASTERISK-05273: h323 error
ASTERISK-05274: iax trunking with ilbc has audio distortion with vm only
ASTERISK-05275: wrong peer name (IP address/port) looked up for devicestate changes during incoming calls
ASTERISK-05276: Timeout reported in queue_log
ASTERISK-05277: [request] [post 1.2] Say number to Brazilian Portuguese (and other similar languages)
ASTERISK-05278: [patch] SIP peer authentication on an external database (RADIUS - LDAP)
ASTERISK-05279: [patch] app_directed_pickup -- Picking up active (non-rining) call causes block.
ASTERISK-05280: [patch] Move automated dial out call files into a "done" directory instead of delete them
ASTERISK-05281: [patch] videosupport option in sip.conf should not be global
ASTERISK-05282: Per README.cdr can set your own extra variables by using Set(CDR(name)=value)
ASTERISK-05283: [patch] API Command GetVar can only retrieve local variables and not global variables
ASTERISK-05284: [patch] REGEX function provides incorrect string to evaluate
ASTERISK-05285: [patch] Screen param not set correctly
ASTERISK-05286: chan_sip doesn't correctly identify peer's name on incoming calls and does not use the options defined in its section in sip.con
ASTERISK-05287: [patch] [post 1.2] use system libgsm where possible
ASTERISK-05288: SIP does NOT change bindaddr when "sip reload"
ASTERISK-05289: System() doesn't behave as expected when shell command returns non-zero
ASTERISK-05290: Typo in function name destroy_odbc_obj
ASTERISK-05291: [PATCH] manager documentation incorrect for passing channel variables
ASTERISK-05292: [patch] app_disa does not properly jump to the "i" extension if an invalid extension was received.
ASTERISK-05293: [patch] [post 1.2] STAT()
ASTERISK-05294: ASTCC - Crashes on free calls. -- Patch Included
ASTERISK-05295: Chanspy ruin the voice quality if any of the 3 party is using IAX
ASTERISK-05296: Asterisk crashes trying to use a Cisco 7920 phone
ASTERISK-05297: [patch] pbx_builtin_setvar_helper and __ prefixed variables
ASTERISK-05298: [patch] Depmod failure while installing on different kernel version
ASTERISK-05299: [patch] Allows for easy building of v1-0 zaptel on a different kernel version
ASTERISK-05300: [patch] GotoIf() evaluates wrong
ASTERISK-05301: [patch] small performance tweak in ast_variable_retrieve
ASTERISK-05302: Misc bugs in manager api
ASTERISK-05303: [patch] Fix usage of variadic function
ASTERISK-05304: [patch] [post-1.2] QUEUE_MEMBER_LIST
ASTERISK-05305: GROUP_COUNT inconsistencies
ASTERISK-05306: Unable to re-load chan_iax2
ASTERISK-05307: [GCC4.1][1.0.9] local variable used before set
ASTERISK-05308: Read() falls through if ResponseTimeout() is reached but fails to land in 't' (timed out)
ASTERISK-05309: [patch] Fix some double unplus good grammar
ASTERISK-05310: Cannot compile h323 on FreeBSD
ASTERISK-05311: [patch] Off by 1 issue in SORT()
ASTERISK-05312: [patch] Don't compile app_page without meetme
ASTERISK-05313: [patch] Update help text of disa() - add "i"
ASTERISK-05314: Crash when answering call and using queue monitoring
ASTERISK-05315: [patch] README update
ASTERISK-05316: [patch] Update CREDITS
ASTERISK-05317: [patch] Formatting of README.sms
ASTERISK-05318: [patch] SECURITY update
ASTERISK-05319: [patch] Update asterisk man page (asterisk.sgml)
ASTERISK-05320: Read seems to stop execution in the dial plan if 'User disconnected' is reached.
ASTERISK-05321: [feature request] detect if any agents are available without answering call before first answer, and reject of not.
ASTERISK-05322: [patch] Variable initialization
ASTERISK-05323: [patch] GSM - Remove unused variables
ASTERISK-05324: [patch] README.ael update
ASTERISK-05325: Ringing multiple sip phones gives warning
ASTERISK-05326: [patch] Add Max-Forwards to all requests
ASTERISK-05327: [patch] 100 trying after BYE/200 OK keeps session alive for ever and ever
ASTERISK-05328: Octal IP address: Asterisk bug?
ASTERISK-05329: octal interpretation
ASTERISK-05330: make install fails for ooh323
ASTERISK-05331: Typo/cut'n'paste error in app_dial *descrip
ASTERISK-05332: DeadAGI script terminates when hangup during STREAM FILE or SAY NUMBER
ASTERISK-05333: Extensions.conf error
ASTERISK-05334: Portability issues with FreeBSD - OOH323
ASTERISK-05335: [patch] Kill sessions that never start
ASTERISK-05336: [branch] Support for Diversion: header in redirects of calls with 302
ASTERISK-05337: h323 show peers loop forever
ASTERISK-05338: chan_sip forgets IP address of dynamic client when registration expires
ASTERISK-05339: zap channels not closing after hangup
ASTERISK-05340: driver not available in normal way
ASTERISK-05341: recent changes to utils.c seg. faults asterisk
ASTERISK-05342: No music on hold after passing trough chan_agent
ASTERISK-05343: [patch] columns in "zap show status" do not line-up
ASTERISK-05344: 2.6 driver assumes 1000 HZ clock
ASTERISK-05345: Adjustments to Makefile and sources to assist in compilation on cygwin platforms
ASTERISK-05346: Asterisk fails to return stale=true when a nonce is stale
ASTERISK-05347: [Patch] vm_newuser incorrectly handles password change
ASTERISK-05348: Inbound PRI call not handle if called SIP exten is registered but offline
ASTERISK-05349: [patch] [post 1.2] Add new channel variable FORWARD_CONTEXT and implement it in app_dial.c
ASTERISK-05350: [patch?] Every second reload of cdr_odbc will produce an error.
ASTERISK-05351: Variables lost during attended transfer
ASTERISK-05352: AgentCallbackLogin parsing issues
ASTERISK-05353: [branch] AstJAB: res_jabber && chan_jingle
ASTERISK-05354: [patch] segfault in rtp.c:345
ASTERISK-05355: [patch] Update coding guidelines
ASTERISK-05356: segfault (in atxfer ??)
ASTERISK-05357: [patch] Improve doxygen docs with README files and configuration files
ASTERISK-05358: slinear inconsistently byteswapped
ASTERISK-05359: Asterisk does not process DTMF when are dialed too fast....
ASTERISK-05360: zapata.conf not reloaded on asterisk reload
ASTERISK-05361: [patch] Fix up AgentCallbackLogin arguments
ASTERISK-05362: Adding #include directory causes Asterisk to deadlock (and in the case of Debian Sarge the entire machine)
ASTERISK-05363: app_muxmon does not work
ASTERISK-05364: IAX to IAX calls to ringing extensions doesn't ring on the near end.
ASTERISK-05365: [patch] More doxygen improvements
ASTERISK-05366: [patch] devicestate fails with '-' in host name
ASTERISK-05367: [patch] [post 1.2] Cannot track device state of MGCP channels
ASTERISK-05368: AgentCallbackLogin doesnt set any agent variables
ASTERISK-05369: [patch] Build error cdr_tds under FreeBSD 7.0
ASTERISK-05370: typo in ast_play_and_record().
ASTERISK-05371: Festival does not generate any sound when I connect with my SIP phone
ASTERISK-05372: [patch] fix for the kb1 echo canceller
ASTERISK-05373: per sip.conf "example" musicclass= is supposed to work for [general] and individual users/peers - it fails for indiv users/peers
ASTERISK-05374: More doxygen cleanup
ASTERISK-05375: [patch] no more strcpy in channels/*.c
ASTERISK-05376: [patch] [post 1.2] sub-architechture-specific directory for modules
ASTERISK-05377: ResetCDR(w) crashes asterisk in case of linked cdrs
ASTERISK-05378: [patch] Carrier ENUM support + some rewrites for ENUMLOOKUP()
ASTERISK-05379: [patch] utils/Makefile fix for Solaris
ASTERISK-05380: [patch] [post 1.2] Preparation for adding support for different sorts of tests into app_test
ASTERISK-05381: [patch] Provide alternative pacing for outgoing SIP registrations to handle 1000s of them
ASTERISK-05382: [patch] add some warning messages to sip thread if its getting bogged down
ASTERISK-05383: [patch] [post 1.2] open up agent variables to the dialplan
ASTERISK-05384: call parking agent channel call lost
ASTERISK-05385: DBPut replacement Set(DB(family/key)=value) not working with variables
ASTERISK-05386: [patch] some little comment clarifications, and improve debug logging
ASTERISK-05387: [patch] [post 1.2] Option to add timestamp to unsolicited manager events.
ASTERISK-05388: Answering a SIP channel, then Dialing and going into a quiet meetme causes a transmit frame type error
ASTERISK-05389: Zaptel-1.0.9.2 channel drivers do not include <asm/io.h> on Alpha.
ASTERISK-05390: Compile-Error in chan_agent.c (latest CVS)
ASTERISK-05391: Dropping extra frame of G.729 since we already have a VAD frame at the end
ASTERISK-05392: [patch] update_registry does things in the wrong order.
ASTERISK-05393: [patch] "don't go backwards" timestamp mangling in calc_fakestamp prevents receiver from correctly restoring wrapped timestamps
ASTERISK-05394: Can't build asterisk because of recent res/Makefile changes.
ASTERISK-05395: wording defining peer/user in sip.conf.sample could be improved.
ASTERISK-05396: mantis security issues...
ASTERISK-05397: call-limit not reset on closed channels
ASTERISK-05398: Broadworks servers apperently (ab)use "OPTIONS" as a status PING, ast should return OK, not 404
ASTERISK-05399: ast_app_getvoice is not writing frames
ASTERISK-05400: Crash in ASTERISK_LIST_INSERT_TAIL
ASTERISK-05401: [patch] getloadavg() not available on uclibc systems
ASTERISK-05402: Can't build asterisk channel H323 with latest CVS (30 Oct 2005)
ASTERISK-05403: Segfault from time to time in todays head
ASTERISK-05404: [patch] chan_agent.c doesn't compile
ASTERISK-05405: Asterisk stops with killed on dialling sip phones
ASTERISK-05406: sip peers are not retained after restart
ASTERISK-05407: Overflow at calculations causes 100% CPU usage
ASTERISK-05408: config.c:524 process_text_line: No '='
ASTERISK-05409: Using cvs-head, unable to bridge or transfer call without crashing.
ASTERISK-05410: Cannot compile against UcLibc-0.9.27
ASTERISK-05411: Syntax Error in Makefile triggered when user doesn't have kernel sources
ASTERISK-05412: [patch] Allow Cut and CUT() to take functions-with-multiple-arguments as its own first argument
ASTERISK-05413: [patch] janitor project - app_mixmonitor - ast_strlen_zero now checks for the string to be defined
ASTERISK-05414: Asterisk crashes upon call transfer finalization
ASTERISK-05415: [patch] ODBC handles become invalid during disconnect/reconnect
ASTERISK-05416: calls to ast_request_and_dial should have a uniform way to inherit variables from a parent channel when necessary
ASTERISK-05417: DISA passwords in file are not validated
ASTERISK-05418: [Patch] chan_misdn updates to compile in cvs head
ASTERISK-05419: Double DTMF sent on T1 to T1 Native Bridge
ASTERISK-05420: [patch] Update severely out of date BUGS file
ASTERISK-05421: Fix warnings on cygwin and fix prototype for getloadavg replacement functiton
ASTERISK-05422: Deadlock debugging code broken
ASTERISK-05423: [patch] Add support for global variables to SetVar Manager API action (to make it work like GetVar)
ASTERISK-05424: [patch] Recent Cygwin portability patches broke cross-compilation
ASTERISK-05425: [patch][post 1.2] Play tone from indications.conf in app_read instead of playing file
ASTERISK-05426: [branch] Find-Me / Follow-me application
ASTERISK-05427: Asterisk 1.2.0 beta 2 is not able to run as a non-root user
ASTERISK-05428: [patch] Extra "klokken" at ast_say_date_with_format_da, format H and k
ASTERISK-05429: [patch][post-1.2] Make app_queue behave a little better
ASTERISK-05430: MeetMe Conferencing Application Locks Sip Channels, and Stop Functioning
ASTERISK-05431: Problem in the Zaptel Makefile for the X86-64 arch
ASTERISK-05432: janitor project - priority jumping and APP_ARGS conversions for pqm,upqm,aqm,and rqm
ASTERISK-05433: [patch] setpriority failing should not be fatal at startup.
ASTERISK-05434: [patch] Documentation about structure of Realtime tables
ASTERISK-05435: Internal HINT state is becoming corrupt/invalid.
ASTERISK-05436: app_mixmonitor don't work with calls that use chan_local
ASTERISK-05437: error parsing multiple variables in Manager Originate
ASTERISK-05438: [patch] Off by 1 issue in 'iax2 debug'
ASTERISK-05439: [patch] Show functions on CLI even when specified in lowercase
ASTERISK-05440: [patch] packetization patch for ooh323c channel driver
ASTERISK-05441: calling GET FULL VARIABLE multiple times in sequence
ASTERISK-05442: Compilation problem with chan_h323 at gcc 3.3.1 and higher
ASTERISK-05443: [patch] [post 1.2] AGI's don't receive variables
ASTERISK-05444: Stops after a couple of system commands
ASTERISK-05445: route by caller id problom
ASTERISK-05446: Multible sip incoming accounts
ASTERISK-05447: lastdigitts is never initialized which causes first DTMF tone to have a random timestamp
ASTERISK-05448: [patch] vm-opts.gsm description in sounds.txt doesn't match actually recorded file.
ASTERISK-05449: [Patch] chan_misdn gives only one way audio
ASTERISK-05450: jumping janitor project
ASTERISK-05451: dnsmanager: iax2 peers with hostnames never come out of UNKNOWN qualify status
ASTERISK-05452: Dial statuses are not well reported to 'parent' channel for local dial:
ASTERISK-05453: [patch] No audio on ATXFER
ASTERISK-05454: Problem with app_dial & cause handling
ASTERISK-05455: runaway chan_local "Not posting to queue" causing Asterisk freeze(defunct)
ASTERISK-05456: Local channels don't inherit _ prefixed variable
ASTERISK-05457: [patch] What to do when on boring flights and trains... Doxygen updates.
ASTERISK-05458: Forwarding ODBC Voicemail Messages Broken
ASTERISK-05459: [patch] filename cannot contain a dot
ASTERISK-05460: Loading *app_muxmon.so* makes Asterisk crash
ASTERISK-05461: muxmon to mixmonitor
ASTERISK-05462: janitor jump priority patch
ASTERISK-05463: [request] Timezone Switch Application
ASTERISK-05464: janitor jump project - app_db
ASTERISK-05465: janitor jump priority patches
ASTERISK-05466: janitor jump priority patch
ASTERISK-05467: [patch] formatting fixes for app_meetme.c
ASTERISK-05468: [patch] Added newline for mixmonitor CLI command
ASTERISK-05469: extra trailing ; in AST_DECLARE_APP_ARGS
ASTERISK-05470: chan_oss: misinitialized variables cause audio to be blocked
ASTERISK-05471: chan_agent won't compile
ASTERISK-05472: [patch] chan_zap does not compile without ZAPATA_PRI
ASTERISK-05473: make running as non-root an asterisk.conf option
ASTERISK-05474: app_externalivr.c - compatibility fix for older gcc versions.
ASTERISK-05475: janitor jumping patch - new app args parsing
ASTERISK-05476: DB Connection in AGI
ASTERISK-05477: jumping janitor patch - some app API additions for arg parsing
ASTERISK-05478: jumping janitor patch - app args parsing patch
ASTERISK-05479: jumping janitor patch
ASTERISK-05480: jumping janitor patch
ASTERISK-05481: AgentCallbackLogin with options
ASTERISK-05482: [patch] application arguments with options are not parsed correctly
ASTERISK-05483: when exiting a conference room, DTMF tones are not properly translated back into the dialplan
ASTERISK-05484: "make dont-optimize" is not currently functional
ASTERISK-05485: [patch] assorted Makefile fixes
ASTERISK-05486: [patch] allow compilation without zaptel
ASTERISK-05487: [post-1.2] [patch] time_t printf/scanf handling
ASTERISK-05488: ast_localtime proposed change
ASTERISK-05489: [patch] [post 1.2] Postgresql Realtime Driver
ASTERISK-05490: jumping janitor patch
ASTERISK-05491: [Patch] fixes for chan_misdn
ASTERISK-05492: jumping janitor patch - and app args parsing conversion
ASTERISK-05493: SJphone "Awaiting acknowledgement" error after updating Asterisk to CVS-HEAD of September
ASTERISK-05494: jumping janitor patch - formatting changes - param options
ASTERISK-05495: jumping janitor patch
ASTERISK-05496: Feature Request: Mailbox Setup for New Users for Voicemail
ASTERISK-05497: ast_expr2.fl Expression Parser Errors
ASTERISK-05498: jumping janitor patch
ASTERISK-05499: jumping janitor patch
ASTERISK-05500: ast_strlen_zero - Janitor project
ASTERISK-05501: jumping janitor patch
ASTERISK-05502: res/res_*odbc.c missing stdio.h
ASTERISK-05503: Can't compile asterisk-addons
ASTERISK-05504: IAX registration lost
ASTERISK-05505: res_config_odbc.c compilation fails
ASTERISK-05506: Severe intermittent reverberation/echo eventually resulting in system crash.
ASTERISK-05507: Strategy roundrobin in ACD Queues
ASTERISK-05508: eyeBeam presence service makes Asterisk crash
ASTERISK-05509: [branch] Patch to allow Dial() to ignore call forward
ASTERISK-05510: Trivial error, include stdio.h
ASTERISK-05511: Multi-Linking and UDP/SIP
ASTERISK-05512: chan_ooh323 doesn't compile with latest CVS
ASTERISK-05513: [patch] asterisk/file.h now depends upon stdio.h
ASTERISK-05514: chan_spy crashes asterisk server
ASTERISK-05515: [patch] Voicemails marked for deletion are not deleted
ASTERISK-05516: removing characters from string problem
ASTERISK-05517: Asterisk fails to start on RedHat ES4 after build.
ASTERISK-05518: [patch][post 1.4] Add NAPTR string parsing to ENUMLOOKUP function call
ASTERISK-05519: [post 1.2] Background Passes Vars to parent context
ASTERISK-05520: [patch] Fix compile on cygwin (proper lock.h changes)
ASTERISK-05521: Segfault in clone_variables that's dependant on console debug setting.
ASTERISK-05522: Voicemail Gain Argument
ASTERISK-05523: [Patch] small patch to avoid warnings while compiling
ASTERISK-05524: [patch] When "autoanswer=yes" channel doesn't auto hangup.
ASTERISK-05525: [patch] Remove deprecated applications StripMSD, Prefix, Suffix
ASTERISK-05526: app_enumlookup is borked
ASTERISK-05527: [patches for review/feedback] [post 1.2] say.c rewrite.
ASTERISK-05528: DISA passwords in file are not validated - fix provided
ASTERISK-05529: Asterisk send BYE to an ext after get 2 invites messages
ASTERISK-05530: [patch] [post-1.2] [app arg] chanisavail janitor
ASTERISK-05531: [patch] [post-1.2] [app arg] CURL() janitor
ASTERISK-05532: If the conference leader is disconnected from a meetme conference that's locked they can not reenter.
ASTERISK-05533: [patch] [post-1.2] [app arg] app_directory janitor
ASTERISK-05534: [patch][post-1.2] Play courtesytone to both parties when call is unparked
ASTERISK-05535: AGI does not set variables until script ends
ASTERISK-05536: When using the D option of the Dial command, DTMF tones not sent correctly
ASTERISK-05537: [patch] deadlock in chan_iax2 on tab complete
ASTERISK-05538: [patch] [post-1.2] [app arg] ExternalIVR janitor
ASTERISK-05539: Music on Hold fpm files are not distributable under the GPL
ASTERISK-05540: [patch] [post-1.2] [app arg] janitor for VMCOUNT() function
ASTERISK-05541: Changing fmt=wav49|gsm|wav to fmt=wav49 in voicemail.conf corrupts mailbox
ASTERISK-05542: [patch] [post-1.2] [app arg] app_meetme janitor
ASTERISK-05543: [request] [post-1.2] New CLI command: show unused extensions
ASTERISK-05544: [patch] whitespaces in AGI
ASTERISK-05545: Cannot compile against uClibc 0.9.27. Several files fail for: "use_ast_mutex_t_instead_of_pthread_mutex_t"
ASTERISK-05546: [patch] [post-1.2] Multiple extensions pickup
ASTERISK-05547: [patch] [post 1.2] realtime dialplan function
ASTERISK-05548: Choppy Audio When Using > 20 MS Audio Frames
ASTERISK-05549: SIP attack through Max-Forwards
ASTERISK-05550: Asterisk doesn't close SIP channel after callee hangs up / "Avoided initial deadlock"
ASTERISK-05551: [patch] memory leak at iax2-provision.c
ASTERISK-05552: [patch] memory leak at pbx_ael.c
ASTERISK-05553: [patch] [post 1.2] adding realtime to MeetMe
ASTERISK-05554: [patch] Get rid of uninitialized warming in res_features.c
ASTERISK-05555: [patch] app_voicemail fails when copying to multiple mailboxes
ASTERISK-05556: [patch] AddQueueMember segfaults Asterisk
ASTERISK-05557: [patch] Memleak in app_rpt
ASTERISK-05558: amd64 / em64t compile error
ASTERISK-05559: init asterisk script bug
ASTERISK-05560: [patch] ast_app_separate_args, not ast_separate_app_args
ASTERISK-05561: Monitor() function broken since beta2 ?
ASTERISK-05562: M option for Macro in a Dial() has a bug
ASTERISK-05563: [patch] SIP Call-Id/tag generation is not unique
ASTERISK-05564: [patch][1.2] app_record leaks frame when failing to write to file
ASTERISK-05565: Compile error in chan_iax2.c, Freebsd
ASTERISK-05566: -L /usr/local/lib needed in Makefiles on FreeBSD
ASTERISK-05567: Logging where incoming calls enters dialplan
ASTERISK-05568: Asterisk 1.2.0-rc1 compilation fail.
ASTERISK-05569: When a blind transfer is initiated if an invalid extension is dialed the caller being transferred is played the invalid message.
ASTERISK-05570: [patch] Forwarding voicemail messages with prepend does not work
ASTERISK-05571: Ztdummy w/ RTC Support Degrades Audio
ASTERISK-05572: automon stopped working
ASTERISK-05573: Resetting password through VoiceMailMain does not update voicemail.conf
ASTERISK-05574: Dial app is not respecting ARGs that use parameters
ASTERISK-05575: Makefile for asterisk-addons needs CFLAGS+=-I../asterisk/include
ASTERISK-05576: thinks channel is in use then receives a call over it.
ASTERISK-05577: wrong include order breaks compilation on freebsd 4.11
ASTERISK-05578: Blind transfer between two Asterisk 1.2rc2 with IAX native bridging fails after 2nd blind transfer
ASTERISK-05579: [patch] [post 1.2] remove a bit of duplicated code and fix some indentation.
ASTERISK-05580: Fedora Core 4 fails to make
ASTERISK-05581: only noise is received over chan_modem_i4l
ASTERISK-05582: app_directory fails to dial realtime user
ASTERISK-05583: buglet in cdr,res/Makefile on FreeBSD
ASTERISK-05584: Apparent SIP deadlock with cdr_pgsql || cdr_odbc
ASTERISK-05585: [patch] add iax.pc for libiax
ASTERISK-05586: [patch] [post-1.2] Give app_realtime status variables
ASTERISK-05587: issuing a reload several times at cli prompt crashes asterisk
ASTERISK-05588: [patch] app_disa broken in 1.2rc2
ASTERISK-05589: Passing variables using originate command in Asterisk Manager Interface causes seg fault
ASTERISK-05590: [patch] asterisk-sounds "make update"
ASTERISK-05591: [Patch]: chan_misdn: minor bugfixes, regarding applications
ASTERISK-05592: [ooh323c] Channel Locking
ASTERISK-05593: [patch][post 1.2] Report dialplan reload in manager
ASTERISK-05594: [patch][post 1.2] Report SIP reload in manager
ASTERISK-05595: [patch] Update README.variables with status variables
ASTERISK-05596: reload or reload pbx_ael.so does not reload the dialplan from extensions.ael
ASTERISK-05597: compile time error formats format_ogg_vorbis.c
ASTERISK-05598: AgentCallbackLogin Ver 1.2 rc2
ASTERISK-05599: * Sends 403 Unauthorized upon reception of INFO method from a Nortel MCS 5200 sip proxy
ASTERISK-05600: "make install" kills /var making system unusable
ASTERISK-05601: randomly interrupted playback (ast_waitstream)
ASTERISK-05602: [Patch] [post 1.2] Meetme participant duration tracking
ASTERISK-05603: Flash events is not sent correctly as sip event
ASTERISK-05604: [patch] app_playback option parsing's inverted
ASTERISK-05605: Asterisk crash on heavy load using AGI scripts
ASTERISK-05606: execution of manager's command "hangup", when agent receive voice packets, cause deadlock
ASTERISK-05607: Problem on routing a call between Asterisk-CVS-HEAD(from 18. Oct. 05) and Asterisk-1.2-rc2
ASTERISK-05608: [post 1.2] Realtime fails using unixODBC and FreeTDS against MSSQL
ASTERISK-05609: 1.2rc2 fails to build correctly on MacOS X 10.4.3
ASTERISK-05610: chan_oss no audio path - one way audio path
ASTERISK-05611: [post-1.2] unused code
ASTERISK-05612: seg fault if channel variable in Setvar is not set
ASTERISK-05613: [patch] equal sign in application argument confuses AEL
ASTERISK-05614: Agent calls not recorded after transfer
ASTERISK-05615: Asterisk crashes if ForkCDR is used in macro called from Dial()
ASTERISK-05616: SIP - H263: Found video format unknown
ASTERISK-05617: [post-1.2] Avoid SQL syntax in realtime core functions
ASTERISK-05618: res_musiconhold.c: Music on Hold class 'default' already exists
ASTERISK-05619: [patch] Document that FreeTDS is not supported for realtime in 1.2
ASTERISK-05620: [branch][post 1.4] LDAP Realtime driver
ASTERISK-05621: chan_sip.c: Autodestruct on call '' with owner in place -> CRASH
ASTERISK-05622: make install places drivers in wrong directory
ASTERISK-05623: [patch] Zaptel Makefile Changes
ASTERISK-05624: [patch] X option of MeeMe doesn't work
ASTERISK-05625: [patch] VoiceMail() in the dial plan segfaults after collecting an extension
ASTERISK-05626: "make install" fails on Solaris for 1.2.0
ASTERISK-05627: in commet field of config file double minus ends parsing of the config file ";--"
ASTERISK-05628: [patch] Goto with parens does not compile correctly.
ASTERISK-05629: Asterisk crashes after issuing "End MixMonitor Recording" message
ASTERISK-05630: [branch] Devicestate for chan_local and parking
ASTERISK-05631: Asterisk sends invalid SDP, causes RFC2833 not to work.
ASTERISK-05632: Segmentation Fault at DISA
ASTERISK-05633: DTMF detection via GSM and VoIP very unreliable
ASTERISK-05634: Asterisk 1.2 frequent crashes in queue_destroy
ASTERISK-05635: Asterisk crashes if zapata.conf no exist or configured for not instaled hardware
ASTERISK-05636: [patch] RedHat / Debian Init Script Updates for 1.2
ASTERISK-05637: [patch] Remove outdated redhat/asterisk script
ASTERISK-05638: [patch] memory leak in channel.c
ASTERISK-05639: [patch] ast_osp_lookup does not pass "numresults" to ast_osp_next
ASTERISK-05640: Deadlock on 'show translation'
ASTERISK-05641: Cosmetic issue - func_enum.so
ASTERISK-05642: [patch] fixes for UPGRADE.txt and 'iax2 show peer <foo>'
ASTERISK-05643: [patch] code simplification
ASTERISK-05644: lots of these: chan_sip.c:11253 do_monitor: chan_sip: ast_sched_runq ran 70 all at once
ASTERISK-05645: [patch] app_hasnewvoicemail incorrectly reports priority n+101 does not exist
ASTERISK-05646: [patch] make fails on Debian/testing
ASTERISK-05647: dialplan will load all matching files starting with extensions.conf
ASTERISK-05648: Getting 'detailed' cause instead of only DIALSTATUS
ASTERISK-05649: chan_oss removed from the build ?
ASTERISK-05650: [patch] add callerid feature to chan_oss
ASTERISK-05651: callerid feature for chan_oss
ASTERISK-05652: argument error to ast_copy_string in res_musiconhold.c
ASTERISK-05653: [patch] cli.c - remove some useless calls to ast_strlen_zero
ASTERISK-05654: [patch] pbx.c - remove some unnecessary checks
ASTERISK-05655: Malformed callfile crashes server
ASTERISK-05656: Major problem with queues in 1.2.0
ASTERISK-05657: [patch] Text conferencing
ASTERISK-05658: Deadlock in app_queue
ASTERISK-05659: [patch] Local channel variable behavior - variable inheritance
ASTERISK-05660: AGI STREAM FILE does not seem to be working for mp3 files even when format_mp3 is installed
ASTERISK-05661: does not go to most specific pattern
ASTERISK-05662: random Crashs when using Monitor on SIP extensions
ASTERISK-05663: iax2_getpeername res=1 missing
ASTERISK-05664: res=1 missing in function iax2_getpeername
ASTERISK-05665: [patch] add WITHOUT_PRI Makefile knob
ASTERISK-05666: SIP useragent needs to be on a per-registration basis
ASTERISK-05667: H323 calling feature
ASTERISK-05668: random Crashes related to chan_sip
ASTERISK-05669: meetme list tab completion broken
ASTERISK-05670: NCS 1.0 feature support request
ASTERISK-05671: deprecated BYE/Also transfer method
ASTERISK-05672: [patch] Monitor() recordings have 12dB too much gain
ASTERISK-05673: Missing SIP status codes
ASTERISK-05674: [patch] System name option and variable
ASTERISK-05675: Inconsistency between ringcadence and ringcadance
ASTERISK-05676: Meetme app ROBOTIC SOUND
ASTERISK-05677: [patch] users change their mailbox password, and these changes are lost after an asterisk reload
ASTERISK-05678: [patch] Cosmetic issue -- "yay!" printed on console if a voicemail file found.
ASTERISK-05679: [patch] app_curl - ability to set proxy
ASTERISK-05680: Customizable Fields on SQLite CDR Backend
ASTERISK-05681: [patch] capture rfc2833 events in the 1.0 tree
ASTERISK-05682: #exec does not work in all instances
ASTERISK-05683: [branch] sip_register() cleanup and implement "register => peername" command
ASTERISK-05684: MP3 module starts mpg123 processes and never kills them
ASTERISK-05685: MP3 module starts mpg123 processes and never kills them
ASTERISK-05686: Called party transfer does not work if the call arrives via IAX and is transfered out via IAX
ASTERISK-05687: RFC2833 DTMF detection not working on incoming PSTN->SIP->* calls (not bug 5780!)
ASTERISK-05688: Indicating H option does not accept * to hangup if call is not connected
ASTERISK-05689: [branch] Bridge two channels via a Dialplan App or an AMI event
ASTERISK-05690: ASTCC AGI get_variable can not get?
ASTERISK-05691: Calls using G.729 (with license) get only static like sound.
ASTERISK-05692: Make channel list (show channels) available via Manager API
ASTERISK-05693: Make channel list (show channels) available via Manager API
ASTERISK-05694: deadlock errors
ASTERISK-05695: asterisk hangup
ASTERISK-05696: [patch] Originate CLI command
ASTERISK-05697: [patch] Zaptel compilation fails on 2.6.15-rc2-git6
ASTERISK-05698: [patch] Tone data for Argentina
ASTERISK-05699: [patch] wrong argument to ast_copy_string
ASTERISK-05700: DTMF does not working
ASTERISK-05701: Cannot install ooh323c
ASTERISK-05702: [patch] Alternative SIP call pickup with Caller ID displayed
ASTERISK-05703: [patch] Allow ',' in extconfig database and table
ASTERISK-05704: [patch] dialplan hint system is case sentive
ASTERISK-05705: [patch] constification of ast_var_name(), ast_var_value(), pbx_builtin_getvar_helper() and friends.
ASTERISK-05706: R2 Support for Argentina
ASTERISK-05707: [patch][post 1.4] app_dial code restructuring.
ASTERISK-05708: Truncated CDR records
ASTERISK-05709: DialedTime variables always 0 if call is not answered
ASTERISK-05710: [patch] queue weight feature is broken
ASTERISK-05711: SIP SDP message of Answer() on a G729 channel prevents DTMF from working
ASTERISK-05712: [patch] duplicate list of variable names in cdr.c
ASTERISK-05713: AgentCallbackLogin and SIP hold music doesn't work in user to agent direction.
ASTERISK-05714: Embedded SIP reserved characters may cause indigestion to cdr-csv
ASTERISK-05715: [patch] assorted bug and performance fixes for res_agi.c
ASTERISK-05716: [patch] SIP provider sip.qsc.de responds with multiple-line header in "WWW-Authenticate"
ASTERISK-05717: [patch] voicemail change password glitch
ASTERISK-05718: answer delay
ASTERISK-05719: [patch] CUT() function has no error msg
ASTERISK-05720: "CUT" function does not seem to function as expected, or at all
ASTERISK-05721: Add "fromdomain" description in sip.conf
ASTERISK-05722: [patch] make update fails with subversion repository
ASTERISK-05723: channel variables in context templates not available
ASTERISK-05724: EM Wink problems
ASTERISK-05725: [POST 1.2] Add UserAgent to astdb
ASTERISK-05726: [patch] chan_agent.c seg faults
ASTERISK-05727: svn revision missing from show version if built in non english locale
ASTERISK-05728: [patch] Allow use of separate MySQL servers for SELECT and UPDATE queries
ASTERISK-05729: [patch] app_festival will stay indefinitly in while loop if Festival server goes down
ASTERISK-05730: [patch] fix build error
ASTERISK-05731: [patch] Add 'From:' field to manager ParkedCalls command
ASTERISK-05732: meetme with i option - high cpu usage scenario
ASTERISK-05733: [patch] bug in build_tools/make_svn_branch_name
ASTERISK-05734: BYE instead of response causes stuck channel
ASTERISK-05735: [patch] Able to access other voicemail boxe if entering VoicemailMain() with no context
ASTERISK-05736: [patch] Page() does not support passing ALERT_INFO variable.
ASTERISK-05737: [patch] keypad facility parsing / libpri_copy_string problem
ASTERISK-05738: all files in svn are executable
ASTERISK-05739: [request] T1/A timers tweakable on a per-peer basis
ASTERISK-05740: If using G729 as Codec, Asterisk adds an option line, telling CPE to disable annexb
ASTERISK-05741: error in cdr_addon_mysql
ASTERISK-05742: Show Channels cut some word
ASTERISK-05743: ControlPlayback playback position
ASTERISK-05744: Regitrations do not get removed after a reload
ASTERISK-05745: [patch] Restore context searching via a config file option
ASTERISK-05746: Watchers grows indefinetely while resetting phones with BLF lines
ASTERISK-05747: G.729a Refuses to work with Asterisk 1.2.0 and SVN-Branch-1.2-r7231
ASTERISK-05748: Manager "Agents" action reports AGENT_IDLE for agents on calls
ASTERISK-05749: [new app] HTTPPlayback
ASTERISK-05750: Passing Variable: parameter in Originate crashes CVS-HEAD 2005-11-18
ASTERISK-05751: Originate doesn't set variables with Set
ASTERISK-05752: [patch] ARRAY function
ASTERISK-05753: DTMF-event can be two characters in debug
ASTERISK-05754: Zaptel: undefined symbols rtc_unregister, rtc_control, rtc_register
ASTERISK-05755: Asterisk does not handle MWI notifications from external voicemail systems.
ASTERISK-05756: chan_oss no audio output
ASTERISK-05757: [patch] USing qualify=yes - connection eventually reports UNREACHABLE and never recovers
ASTERISK-05758: L Option hangs up call with more time that specified.
ASTERISK-05759: Problem negotiating mid-call codecs between two gateways having * in the loop
ASTERISK-05760: Problem negotiating mid-call codecs between two gateways having * in the loop
ASTERISK-05761: [patch][post 1.4] support for a better manager output in action iax show peers
ASTERISK-05762: dial-out (pbx_spool) and variable inheritance problems
ASTERISK-05763: [patch] Disposition showing FAILED even though call is answered successfully
ASTERISK-05764: [patch] Hangup on early session
ASTERISK-05765: [patch] Coreless Timeout and Extended Timeout Capability
ASTERISK-05766: [patch] Channel hangs after second ForkCDR+SetCDRUserField
ASTERISK-05767: [patch] dynamic members gets lost with realtime queues
ASTERISK-05768: [patch] show functions like FOO
ASTERISK-05769: [patch] BASE64_ENCODE and BASE64_DECODE
ASTERISK-05770: [patch] Unknown number is not announced properly when reading envelope
ASTERISK-05771: [patch] Chat-cord: Off Hook + DTMF detection
ASTERISK-05772: [patch] Two fields mixed up in error message in sip_xmit
ASTERISK-05773: [patch] Japanese Caller ID retrieval for TDM400P
ASTERISK-05774: [patch] make voicemail files group writable
ASTERISK-05775: [patch][post 1.4] remove dependenices on res_adsi.so in app_voicemail.so and others
ASTERISK-05776: Enable DTMF proxying via MeetMe
ASTERISK-05777: 'info' indication for North America is incorrect
ASTERISK-05778: [patch] [post 1.2] Say syntax in Russian (fit to #5675)
ASTERISK-05779: Calls that come in over Zap channel to SIP phone cannot be "unholded"
ASTERISK-05780: DISA doesn't work over a ISDN PRI Zap channel (timeout waiting for CONNECT ACKNOWLEDGE)
ASTERISK-05781: [patch] Cannot AddQueueMember on realtime queue, if queue not yet loaded
ASTERISK-05782: SIP CANCELs don't seem to be retransmitted per RFC2543
ASTERISK-05783: just testing ...
ASTERISK-05784: 'C' flag dont work properly
ASTERISK-05785: [patch] improve AGI console output
ASTERISK-05786: chanspy() in dialplan causing crash
ASTERISK-05787: Asterisk randomly crashes when using ACD, agents and SIP channels.
ASTERISK-05788: [Patch] update channels/Makefile for current chan_misdn version
ASTERISK-05789: originate action causes server to drop tcp session
ASTERISK-05790: [post 1.2][patch] SMDI message desk support for Asterisk
ASTERISK-05791: chan_h323 dead lock
ASTERISK-05792: channels stuck open...memory leak
ASTERISK-05793: Limit on number of simultaneous app_ices instances
ASTERISK-05794: app_voicemail crashes when both Realtime voicemaand ODBC_STORAGE are enabled
ASTERISK-05795: [patch] "logger rotate" does not rotate out queue_log.
ASTERISK-05796: [patch] uninitialized variable
ASTERISK-05797: [patch] ast_osp_lookup & ast_osp_next do not work correctly without destination protocol information
ASTERISK-05798: random crash in pri_fixup_principle
ASTERISK-05799: [patch] printf-like append_history and use a tailq to store records
ASTERISK-05800: [patch] removal of redundant/repeated code and formatting cleanup
ASTERISK-05801: AgentLogin/AgentCallbackLogin not work with SIP-channel
ASTERISK-05802: variables problem call_forward
ASTERISK-05803: Transfer from Polycom 600 to Zap causes Internal Server Error
ASTERISK-05804: [patch] Answering machine detection
ASTERISK-05805: Recent changes to app_voicemail breaks realtime voicemail
ASTERISK-05806: [patch] new RAND() function
ASTERISK-05807: Transferred phone calls from AgentCallbacklogin drop 50% of the time.
ASTERISK-05808: FSK/DTMF callerid INDIA
ASTERISK-05809: new app_groupcount.so causes CLI name clash; crashes server
ASTERISK-05810: [patch] List management bug in AST_LIST_REMOVE
ASTERISK-05811: error counter are not implemented
ASTERISK-05812: Defining searchcontexts=yes in voicemail.conf generates a crash
ASTERISK-05813: monmp3 thread error
ASTERISK-05814: CUT( ) funtion with field delimiter = comma
ASTERISK-05815: [patch] RFC-2833 DTMF support is broken in Asterisk 1.2.x
ASTERISK-05816: NULL context in VoiceMailMain dialplan command
ASTERISK-05817: New Feature: Make a call from the CLI
ASTERISK-05818: SIP reinvite and realtime peers
ASTERISK-05819: Crash in ast_do_masquerade when forwarding call externally
ASTERISK-05820: [FEATURE] Array Variable handling added
ASTERISK-05821: [PATCH] update for README.misdn
ASTERISK-05822: unclear/useless code near the end of chan_sip.c::handle_request_invite()
ASTERISK-05823: [patch] 'r' option in app_dial not working correctly
ASTERISK-05824: Problem with ManagerAPI events following a Parked Call related events
ASTERISK-05825: ManagerAPI Event - CallerID is set wrong on call (dial)
ASTERISK-05826: Channels being apparently locked by chan_h323
ASTERISK-05827: unique signature do not match beetween queue_log and recorded calls
ASTERISK-05828: Asterisk blocks when a long call causes Monitor() to generate a 2GB file.
ASTERISK-05829: [patch] channel/Makefile WITHOUT_PRI compile hook
ASTERISK-05830: [patch] move duplicated code to a function
ASTERISK-05831: Segmentation fault
ASTERISK-05832: [patch] Append option to record not working
ASTERISK-05833: Attempting to place a call from a queue into a parking extension crashes asterisk (chan-sccp)
ASTERISK-05834: [patch] Bug in wctdm.c
ASTERISK-05835: chanspy only able to spy from one channel
ASTERISK-05836: [patch] Routing based on the Diversion SIP Header REASON code
ASTERISK-05837: Festival doesn't work folling a "Ringing" (Asked to transmit frame type 64, while native formats is 4)
ASTERISK-05838: stub functions for res_monitor, res_features, and res_adsi
ASTERISK-05839: crash whilst attempting a directed call pickup using app_pickup (possible cdr issue)
ASTERISK-05840: Asterisk Crash
ASTERISK-05841: I don't want ilbc, i just want G.711...
ASTERISK-05842: v1.2.1 requires new config setting for voicemail ARA to work
ASTERISK-05843: [post 1.4] improper handling of contexts with same name
ASTERISK-05844: [patch] clone channel destroyed before manager_event is sent
ASTERISK-05845: Cummulative memory consumption using applications ODBC or MYSQL
ASTERISK-05846: [patch] Memory leakage (RSS growth)
ASTERISK-05847: Allow setting of channel with CDR function
ASTERISK-05848: [patch] chan_iax2 does not reload the context after reload chan_iax2.so for a peer
ASTERISK-05849: app_cut has typo and is missing from make file
ASTERISK-05850: [patch] ChanSpy does not support the complete channel name as spec
ASTERISK-05851: mixmonitor makes the second chanspy on a channel crash asterisk.
ASTERISK-05852: when jitterbuffer=yes DTMF is unreliable on IAX2 links
ASTERISK-05853: Do not work n+101, with lookupblacklist
ASTERISK-05854: AEL silently skipping whole contexts.
ASTERISK-05855: [patch] Allow for context includes in realtime (ARA)
ASTERISK-05856: [64bit] cdr_addon_mysql.so crashes on reload even with new mutex code
ASTERISK-05857: AEL - missing paren silently aborts parse of extensions.ael.
ASTERISK-05858: ztdummy missing
ASTERISK-05859: [patch] program for testing "is there any zaptel device?"
ASTERISK-05860: [patch] Forced bypass switch in pbx.c - makes Realtime (or other switch) master
ASTERISK-05861: [patch] possible incomplete locking in sip_pvt's packets list
ASTERISK-05862: [patch] implement AEL via bison/flex.
ASTERISK-05863: [patch] Port of codec libs to MSVC, + fix some bugs and warnings in those libs.
ASTERISK-05864: [patch] RTC support on x86_64 in ztdummy
ASTERISK-05865: Realtime voicemail contexts broken in 1.2.1
ASTERISK-05866: [patch] two duplicated statements in chan_sip.c and a comment
ASTERISK-05867: NoCDR() and agi->exec('SetAMAFlags', 'omit') used together cause Asterisk to crash
ASTERISK-05868: [patch] picking up multiple DTMF using RFC2833
ASTERISK-05869: We have random crashes in less than 30 minutes.
ASTERISK-05870: iax/sip status sent to a proc
ASTERISK-05871: A option in dial function in dialplan looks for wrong filename
ASTERISK-05872: ChanIsAvail application always jumps
ASTERISK-05873: Asterisk crashes when it attempts to free a bogus frame
ASTERISK-05874: Small cummulative memory consumption using application MYSQL
ASTERISK-05875: 3-way conferences with SNOM phones and a Zaptel channel do not work
ASTERISK-05876: [patch] Asterisk sends 403 instead of 401 after receiving auth data with incorrect password
ASTERISK-05877: Make func_enum to use default zone suffixes from enum.conf
ASTERISK-05878: Support for greek syntax
ASTERISK-05879: MixMonitor outputing recording of zero file size
ASTERISK-05880: [patch] agent not auto logoff
ASTERISK-05881: Called number problem
ASTERISK-05882: [patch] AgentCallbackLogin CLI help is displaying wrong parameters
ASTERISK-05883: [branch] asterisk/team/group/autoconf_and_menuselect
ASTERISK-05884: Asterisk crash with voicemail realtime
ASTERISK-05885: Anonymous Caller ID'S
ASTERISK-05886: Dropping extra frame of G.729 since we already have a VAD frame at the end
ASTERISK-05887: [patch] ATF - Asterisk Testing Framework
ASTERISK-05888: [request] Prevent agents from using extension already in use
ASTERISK-05889: [patch] 'Reload' clears SIP subscriptions
ASTERISK-05890: 1.2.1 crashes right after voicemail hangup
ASTERISK-05891: Fix for bug 5427 breaks video in SIP (I think)
ASTERISK-05892: [patch] Voicemail storage with ODBC and PostgreSQL fails due to bad SQL queries.
ASTERISK-05893: [patch] VMCOUNT() returns 0/1 and not number of messages
ASTERISK-05894: Speex and iLBC have the same rtpmap when speex only is offered
ASTERISK-05895: [patch] new command, show threads
ASTERISK-05896: [patch] socket_read(), trying to use frame that was free().
ASTERISK-05897: Manage full INVITE URI
ASTERISK-05898: queue_log is not complete for some actions
ASTERISK-05899: [patch] Deprecate builtin variables that have been replaced with functions
ASTERISK-05900: Crash in meetme: *** glibc detected *** double free or corruption (!prev): (0x......)
ASTERISK-05901: useless code ?
ASTERISK-05902: upgrade to 1.2.1 - voicemail cannot find mailbox in default context, using realtime with mysql
ASTERISK-05903: [patch] odbcstorage + voicemail + !mysql = no good
ASTERISK-05904: [patch] file format conversion CLI command
ASTERISK-05905: [Feature Request] Google Talk channel (chan_gtalk)
ASTERISK-05906: Some Zaptel's failures while working with CONFIG_ZAPATA_NET defined.
ASTERISK-05907: [patch] canonicize ast_walk*() loops
ASTERISK-05908: [patch] fix for several bugs in pbx_config.c
ASTERISK-05909: ChanSpy() creates recording file with wrong permissions
ASTERISK-05910: [patch] various cli fixes (bugs and performance)
ASTERISK-05911: Asterisk does not handle call from a Cisco IAD correctly
ASTERISK-05912: [patch] memory leak in chan_sip.c:::build_rpid()
ASTERISK-05913: broken rport match in chan_sip.c::check_via()
ASTERISK-05914: [patch] The ast_expr2 facility for $[ ] evaluation leaks memory
ASTERISK-05915: [patch] builtin command list should be alphabetized
ASTERISK-05916: [patch] command completion for 'help' command
ASTERISK-05917: [patch] SL bigger than 100%?
ASTERISK-05918: incorrect documentation for ast_trim_blanks()
ASTERISK-05919: Updated extensions.conf.sample
ASTERISK-05920: [patch] Unable to remove temp greeting when using odbcstorage
ASTERISK-05921: H dial no longer working when calling a local extension invoking music on hold
ASTERISK-05922: Enter and Exit tones sound really bad
ASTERISK-05923: [patch] [memory leak] in res_features.c
ASTERISK-05924: [patch] new events: status of DTMF event
ASTERISK-05925: [patch] don't send 180 and 183 together when progressinband=yes
ASTERISK-05926: Calling MixMonitor twice drives Asterisk into deadlock
ASTERISK-05927: User interface for multi-lingual mailboxes
ASTERISK-05928: [patch]show channels concise output delimiter problem/change
ASTERISK-05929: [patch] cleanup ast_dtmf_stream code and documentation
ASTERISK-05930: chan_h323 causes a crash as the call is bridged
ASTERISK-05931: page calls the caller
ASTERISK-05932: [patch] Add Auto-play option to VoicemailMain
ASTERISK-05933: [patch] Race issue in channel.c involving uniqueint (non-unique UniqueID generated)
ASTERISK-05934: [patch][post 1.4] use PHONE_DIALTONE ioctl instead of writing the dialtone in the select
ASTERISK-05935: [patch] chan_sip doesn't recognize realtime peers by IP address correctly
ASTERISK-05936: null pointer dereference in ast_bridge_call()
ASTERISK-05937: music on hold stopped working after update to asterisk 1.2
ASTERISK-05938: [patch] ast_callerid_parse cleanup
ASTERISK-05939: [patch] asterisk.c::listener() descriptor leak
ASTERISK-05940: [patch] remove duplicated code
ASTERISK-05941: [patch] cleanup of parse_variable_name() and pbx_retrieve_variable()
ASTERISK-05942: [patch] more pbx.c cleanup
ASTERISK-05943: [patch] cleanup of pbx.c time parsing code
ASTERISK-05944: [patch] print a msg on restart/stop when convenient on remote console
ASTERISK-05945: [patch] use linked list macros for channel backends
ASTERISK-05946: [patch] fix locking bug and store hints using linked list macros
ASTERISK-05947: [patch] misplaced lock and store translators using linked list macros
ASTERISK-05948: [attached] rc.suse.asterisk for /etc/init.d
ASTERISK-05949: [patch] handle_show_version_files code clean
ASTERISK-05950: debug output prefixes too much with date/time/file
ASTERISK-05951: [patch] fix unprotected list and use linked list macros
ASTERISK-05952: Call-limit does not function while using queue
ASTERISK-05953: Call-limit does not function while using queue
ASTERISK-05954: [patch] small locking bug and list macro conversion
ASTERISK-05955: [branch] Enhance parking to allow multiple parking lots
ASTERISK-05956: Remote crash when sending special BYE/Also packet
ASTERISK-05957: [patch] Clean up source of chan_agent.c
ASTERISK-05958: [patch] Use list macros in autoservice.c
ASTERISK-05959: Dial command doesnt return properly
ASTERISK-05960: [patch] clean up list handling in image.c
ASTERISK-05961: [patch] Temporary Greeting reminder at login
ASTERISK-05962: [branch] Polycom SoundPoint IP ACD agent feature integration
ASTERISK-05963: [patch] small typo in gosubif
ASTERISK-05964: [patch] Add StopMixMonitor() application and CLI completion to mixmonitor
ASTERISK-05965: [patch] Add "if (option_debug)" in front of LOG_DEBUG logging
ASTERISK-05966: Asterisk 1.2.1 making core dumps sporadically. Seems to be cause by IAX2?
ASTERISK-05967: [patch] Add Abandon Queue Manager Event
ASTERISK-05968: fix pri intense debug messages
ASTERISK-05969: PrivacyManager doesn't set CallerID Number
ASTERISK-05970: [patch] count in show channeltypes
ASTERISK-05971: Cannot use multiple values with func_odbc
ASTERISK-05972: [patch] remove setjmp.h from include/asterisk/channel.h
ASTERISK-05973: [patch] fix inaccurate naming of header containing CallerID Number
ASTERISK-05974: [test-this-branch] Add sendtext manager application
ASTERISK-05975: problems with vm when using an iaxy
ASTERISK-05976: [patch] timeout dial plan loop
ASTERISK-05977: small typo on sip.conf.sample
ASTERISK-05978: [patch] Chinese locale
ASTERISK-05979: [patch] zombie conference when hanging-up on PIN entry
ASTERISK-05980: [patch] certain meetme admin/user menu options cause audio problems
ASTERISK-05981: MeetMe wait until the marked user option causes entrance sound to be played twice upon entering conference.
ASTERISK-05982: [patch] app_chanspy.c record in MP3 and other tweaks
ASTERISK-05983: [patch] changed all copyright 2005 to support 2006 too.
ASTERISK-05984: [patch] uninitialized variables and misplaced checks in format_pcm[_alaw].c
ASTERISK-05985: [patch] Parse error when two rpid's are sent in one header (caller and callee)
ASTERISK-05986: [patch][post 1.4] parking extension selection
ASTERISK-05987: [patch] Fix RPID Handling
ASTERISK-05988: [RFC] LOCAL_USER_ADD / LOCAL_USER_REMOVE semantics ?
ASTERISK-05989: Possible locking problem with agents
ASTERISK-05990: Add ${PARKEDAT} to make parkandannounce actually useful :-)
ASTERISK-05991: option to disable sorting in extensions.conf
ASTERISK-05992: [patch] show channeltype foo
ASTERISK-05993: Caller ID Number not available on Polycom hardware after SVN-trunk-7796
ASTERISK-05994: Incoming calls always show disposition FAILED
ASTERISK-05995: [request] Pass AOC information from Zap <-> IAX
ASTERISK-05996: "Failed to create update thread"
ASTERISK-05997: [patch] fix broken create_addr_from_peer (segfault)
ASTERISK-05998: [patch] va_copy not existing on FreeBSD
ASTERISK-05999: [patch] checking for wrong error value of ast_pthread_create